Traditionally, Public Switched Telephone Network (“PSTN”) telephony systems provided service by utilizing relatively homogeneous, centralized switching infrastructures. These infrastructures were homogeneous in the sense that a single service provider, such as the former Bell System in the United States, utilized a relatively limited, uniform group of telecommunications equipment in a voice-only network that provided “plain, old telephony service” (POTS). These traditional infrastructures were uniform in structure and composition, mainly because they were designed from the top on down. In part because they were centralized, these infrastructures generally had knowledge as to the signal transmission characteristics for every piece of equipment involved in each handled call. Based on this knowledge, the traditional PSTN could make adjustments for end-to-end audio-signal loss, thereby optimizing performance with respect to acoustical audio signal level, audio distortion, and echo. Furthermore, guidance as to the audio-signal loss across various telecommunications device types could be found in various standards and technical guidelines.
For example, in order to reduce the echo signals that were unavoidably present in each transmit path, the echo signals would be carried to the receive path of the line side equipment serving the far-end party and reduced there, based upon an audio-signal loss plan conventionally used by each service provider. The loss plan provided that a predetermined fixed amount of loss would be present in a receive path. The particular amount of fixed loss (e.g., 0 db, 3 dB, 6 dB, etc.) depended upon the type of call: intra-office, intra-exchange (local), intra-LATA (toll), or inter-LATA (toll).
In contrast, modern hybrid telecommunications systems typically must offer interconnectivity between disparate telecommunications networks such as datagram-based networks, the Internet being an example of this, and traditional circuit-switched networks. Additionally, a given network often must handle different types of media concurrently. For example, Voice over Internet Protocol (“VoIP”) systems provide voice telephony over the same networks that handle email, video, and other Internet traffic. Moreover, whereas before there were one or two service providers—that is, local providers and possibly long-distance providers—involved in a particular telephony call, now there can be several service providers involved in handling the media data packets of a given call or session. Finally, each provider's telecommunications network might comprise equipment from many more vendors than before.
A telecommunications system that comprises a business enterprise's network poses additional challenges in optimizing the call quality that is experienced by its users. In such a network, there are telecommunications endpoint devices interconnected with private-branch exchanges and teleconference bridges. To complicate the call-quality management, the audio signals passing through these components often continue on through media gateways to different, globally-reaching, service provider networks. There are techniques for managing the audio signals as they pass through the different components both within and outside of the enterprise network, such as automatic gain control (AGC). These techniques, however, often produce unwanted effects, such as “pumping up” background noise, and often mishandle certain types of signals, such as music-on-hold.
Consequently, the audio-signal loss plan in today's telecommunications networks is significantly more complex to manage than ever before. There are more situations in which the signal amplitude is too low or the noise is too high, or both. Therefore, it would be advantageous to provide a system and method for dynamic end-to-end loss compensation, particularly in an enterprise telecommunications network, with an ability to accommodate the characteristics of the various types of telecommunications devices present.